FFmpeg
af_afir.h
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1 /*
2  * Copyright (c) 2017 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #ifndef AVFILTER_AFIR_H
22 #define AVFILTER_AFIR_H
23 
24 #include "libavutil/common.h"
25 #include "libavutil/float_dsp.h"
26 #include "libavutil/opt.h"
27 #include "libavcodec/avfft.h"
28 
29 #include "audio.h"
30 #include "avfilter.h"
31 #include "formats.h"
32 #include "internal.h"
33 
34 typedef struct AudioFIRSegment {
36  int part_size;
42 
44  int *part_index;
45 
52 
55 
56 typedef struct AudioFIRDSPContext {
57  void (*fcmul_add)(float *sum, const float *t, const float *c,
58  ptrdiff_t len);
60 
61 typedef struct AudioFIRContext {
62  const AVClass *class;
63 
64  float wet_gain;
65  float dry_gain;
66  float length;
67  int gtype;
68  float ir_gain;
69  int ir_format;
70  float max_ir_len;
71  int response;
72  int w, h;
75  int minp;
76  int maxp;
77 
78  float gain;
79 
82  int nb_taps;
85  int one2many;
86 
87  AudioFIRSegment seg[1024];
89 
90  AVFrame *in[2];
93  int64_t pts;
94 
97 
99 
102 
103 #endif /* AVFILTER_AFIR_H */
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
int nb_coef_channels
Definition: af_afir.h:84
int * part_index
Definition: af_afir.h:44
int nb_channels
Definition: af_afir.h:83
Main libavfilter public API header.
AVFrame * block
Definition: af_afir.h:47
int input_offset
Definition: af_afir.h:41
int input_size
Definition: af_afir.h:40
int min_part_size
Definition: af_afir.h:92
float ir_gain
Definition: af_afir.h:68
AVFrame * coeff
Definition: af_afir.h:49
int eof_coeffs
Definition: af_afir.h:80
float dry_gain
Definition: af_afir.h:65
AVOptions.
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
AVFrame * buffer
Definition: af_afir.h:48
void ff_afir_init_x86(AudioFIRDSPContext *s)
Definition: af_afir_init.c:30
AVFrame * output
Definition: af_afir.h:51
AVFrame * sum
Definition: af_afir.h:46
RDFTContext ** irdft
Definition: af_afir.h:53
int coeff_size
Definition: af_afir.h:39
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
RDFTContext ** rdft
Definition: af_afir.h:53
AudioFIRDSPContext afirdsp
Definition: af_afir.h:95
int ir_channel
Definition: af_afir.h:74
#define s(width, name)
Definition: cbs_vp9.c:257
int64_t pts
Definition: af_afir.h:93
AVFloatDSPContext * fdsp
Definition: af_afir.h:96
AVFrame * input
Definition: af_afir.h:50
FFT functions.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
int have_coeffs
Definition: af_afir.h:81
Rational number (pair of numerator and denominator).
Definition: rational.h:58
float max_ir_len
Definition: af_afir.h:70
float length
Definition: af_afir.h:66
int nb_partitions
Definition: af_afir.h:35
void ff_afir_init(AudioFIRDSPContext *s)
Definition: af_afir.c:756
common internal and external API header
AVFrame * video
Definition: af_afir.h:91
float gain
Definition: af_afir.h:78
int fft_length
Definition: af_afir.h:38
int len
float wet_gain
Definition: af_afir.h:64
int * output_offset
Definition: af_afir.h:43
int block_size
Definition: af_afir.h:37
AVRational frame_rate
Definition: af_afir.h:73
internal API functions
int nb_segments
Definition: af_afir.h:88